Adaptive filter unit for being used as an echo canceller

ABSTRACT

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f n , A(t 1 , . . . , t M(fn) )) in the frequency domain; to calculate a transformed second audio signal Y(f n , B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.

TECHNICAL FIELD

The invention relates to an adaptive filter unit, in particular forbeing used as an echo canceller. Furthermore, the invention relates to ahearing device, to a method for performing echo cancelling and to acomputer program for controlling an operation of an adaptive filterunit.

BACKGROUND

In recently developed mobile audio devices, acoustic coupling betweenspeaker and microphone during mobile telephony leads to a decreasingquality of respective audio outputs. Therefore, echo cancellerapplications have been developed. Such echo canceller applications areusually formed as filter units and configured to avoid an acousticoutput signal of the mobile audio device to be transmitted back via themicrophone to an external device that is connected with the mobile audiodevice. Particularly important are echo cancelation applications forclosed rooms, which have the tendency to support long echoes.

Document US 2013/251169 A describes an echo canceller that includes asignal-to-echo ratio calculating unit and a residual echo suppressingunit. The signal-to-echo calculating unit computes a signal-to-echoratio SE(n) indicating a ratio of an echo component to a received signalx(n) from a first residual signal and a second residual signal. Thefirst residual signal is obtained using a filter coefficient sequence ofan update filter, which is obtained up to the previous operation. Thesecond residual signal is obtained using an updated filter coefficientsequence that undergoes a coefficient update which is performed, usingan arbitrary update step size μ(n), on the filter coefficient sequenceof the update filter, which is obtained up to the previous operation.The residual echo suppressing unit suppresses the echo componentcontained in the microphone input signal in accordance with thesignal-to-echo ratio the signal-to-echo ratio calculating unit computes.

It is the object of the present invention to provide an improvedadaptive filter unit for being used as an echo canceller.

SUMMARY

According to a first aspect, the invention relates to an adaptive filterunit, in particular for being used as an echo canceller. The adaptivefilter unit comprises a first filter input, a second filter input, aprocessor and a filter output.

The processor may be arranged and configured to

-   -   receive the first and second electric audio signal;    -   calculate and provide audio estimation data X(f_(n), A(t₁, . . .        , t_(M(fn)))) in the frequency domain by calculating a FFT        transform of the first audio signal A(t), for frequencies        f_(n)=f₁, . . . , f_(N), wherein N is a number of FFT bins, and        with a number of sampling points M(f_(n)) of the first audio        signal A(t);    -   calculate a transformed second audio signal Y(f_(n), B(t)),        formed by a transformation of the second audio signal B(t) into        the frequency domain;    -   calculate a filtered audio signal by subtracting delayed audio        estimation data from the transformed second audio signal,        wherein the delayed audio estimation data may be provided by a        memory unit of the adaptive filter unit, which is arranged to        provide a data exchange with the processor, and wherein the        delayed audio estimation data comprises a frequency dependent        time delay compared to the transformed second audio signal; and    -   calculate a processed audio output signal by transforming the        filtered audio signal into the time domain.

The filter output may be configured to provide the processed audiooutput signal.

The first filter input may be configured to receive a first electricaudio signal, indicative of a first audio signal A(t).

The second filter input may be configured to receive a second electricaudio signal, indicative of a second audio signal B(t).

The first filter input and the second filter input may be a filter bankanalysis including time to frequency conversion of the first electricaudio signal and the second electric audio signal, respectively.

A filter bank analysis may be an array of filters, such as a bandpassfilters, that decompose the electric audio signals into multiplecomponents, each component carrying a single frequency sub-band of theoriginal electric audio signal. The output of analysis may be referredto as a subband signal with as many subbands as there are filters in thefilter bank.

The filter output may be a filter bank synthesis including frequency totime conversion of the processed audio output signal.

Another application of filter banks may be signal compression, when somefrequencies are more important than others. After decomposition, theimportant frequencies can be coded with a fine resolution. Smalldifferences at these frequencies are significant and a coding schemethat preserves these differences must be used. On the other hand, lessimportant frequencies do not have to be exact. A coarser coding schemecan be used, even though some of the finer (but less important) detailswill be lost in the coding.

An advantage of the adaptive filter unit is that the time delay isfrequency dependent, which allows signal processing resources like amemory of the echo canceller to be used in the most important parts ofthe frequency spectrum. For instance, the adaptive filter unit couldallow a large time delay, i.e. long filters, for low frequencies, whererooms have the tendency to have the longest echoes. Thus, the adaptivefilter unit according to the first aspect of the invention allows largetime delays for critical frequencies to achieve a good filteringperformance and short time delays for other frequencies to maintain abest use of available processing resources.

Different frequency dependencies of the time delay might be used withinthe adaptive filter unit. Thereby, the adaptive filter unit can beadapted to different applications for a large variety of hearingdevices, such as headsets, headphones and hearing aids. The use inhearing aids is particularly advantageous, since the frequency dependenttime delay can be adapted with respect to the hearing capability of auser of the hearing aid.

Using first and second filter inputs and a filter output is rather usualin the field of echo cancellation. Therefore, the adaptive filter unitcan be advantageously integrated in well known hearing devices by simplyexchanging a prior art adaptive filter previously used in the hearingdevice.

The used Fast Fourier Transform (FFT) provides a well known, simple andfast algorithm to transform the time dependent audio signals A(t) andB(t) into the frequency domain. It is furthermore well known that thesampling points of a signal to be transformed are formed by a number ofdata points of the signal received at respective discrete time points ofthe signal. These sampling points are than used to calculate a datapoint for a single one of the FFT bins in the frequency domain via theFFT. By repeating the transformation to discrete FFT bins for the sameor different sampling points of time dependent in data, FFT transformeddata in the frequency domain is provided.

The data exchange between the processor and the memory unit does atleast provide the audio estimation data to the memory unit and thedelayed audio estimation data from the memory unit back to theprocessor. The delayed audio estimation data is thereby formed by audioestimation data that is stored in the memory unit and therefore delayedcompared to more recent audio signals received by the adaptive filterunit.

The memory unit and the processor may be formed as separated units ofthe adaptive filter unit, and may be integrated into a common unit.

The adaptive filter unit according the first aspect of the inventionwill be described.

The processor of the adaptive filter unit comprises a multidelay filterstructure. The multidelay filter structure provides a block-basedfrequency domain implementation of the well known least mean squarefilter algorithm. This structure partitions a long filter into manysubfilters so that the FFT transforms less data in parallel steps, whichreduces memory requirements and a time delay caused by the processing ofthe processor. A detailed description of the well known multidelayfilter structure is given in J.-S. Soo, K. Pang “Multidelay blockfrequency domain adaptive filter” (IEEE Transactions on Acoustics,Speech and Signal Processing, vol. 38, no. 2, pp. 373-376, 1990).

The memory unit comprises a delay line structure comprising at least onedelay line, which contains at least the audio estimation data X(f_(n),A(′t₁, . . . , t′_(M(fn)))) for points in time t′₁, . . . , t_(M(fn))′)being before the time t₁. The memory unit stores audio estimation datathat has been obtained in the past and thus comprises a time delay withrespect to the time t₁ of a data point of the most recent audio signal.Thus, the memory unit provides a simple structure that allows a storingof data with a respective time delay.

The memory unit comprises a plurality of circular delay lines allocatedto each frequency f_(n), respectively, wherein the circular delay linesvary in line length, which is defined as a maximum number of values thatcan be stored in the respective circular delay line. The circular delaylines advantageously allow a respective delay line for each FFT bin andtherefore a line length that may be adapted to the frequency dependenttime delay. Since the position of a value in the circular delay linedepends on the time delay of the corresponding signal with respect tothe most recent signal, the delay of the value can be computed veryefficiently by the processor or by the memory unit. A varying linelength of the circular delay lines is particularly advantageous for anadaptive filter unit that comprises a fixed frequency dependent timedelay, since the delay line structure does not need to be adapted todifferent frequency dependencies in this case. In circular delay lines,old values are simply overwritten by new values after a predefined timeperiod, which depends on the line length, so that the circular delaylines furthermore avoid the single processing step of deleting values inthe delay lines.

The data exchange between the processor and the memory unit comprises anallocation of a value of the audio estimation data X(f_(n), A(t₁, . . .t_(M(fn)))) for each frequency f_(n) to a memory address of a memorystructure of the memory unit. In a variant, an allocation informationindicative of the respective allocation is received by the processor.The exchange of the allocation of a value can lead to a simple and fastaccess of the processor to delayed data with a well defined time delay.Preferably, the allocation information further comprises a signal timeindicative of the time at which the value of the audio estimation datahas been allocated to the memory address.

Each delay line comprises a line length that may be limited to powers oftwo. In view of the usual binary structure of the memory address, theline length avoids a wasting of memory.

The memory unit comprises a base address register allocating eachcircular delay line to a base address, and further comprises at leastone bookkeeping buffer, configured to store the line length and the baseaddress of each circular delay line. Preferably, separate bookkeepingbuffers may be used for storing the line length and the base address ofeach circular delay line. The use of base address register in general iswell known and easy to integrate in modern signal processing systems. Bystoring the base address, the bookkeeping buffers allow a placing of thecircular delay lines at any memory address. Thus, the bookkeepingbuffers lead to a very efficient usage of the memory and further helpsto provide the best possible use of available resources. The bookkeepingbuffers further remove the need for placing the delay lines at certainplaces in memory. In a variant, the bookkeeping buffer may be used toallow a placing of all delay lines in a block of memory within thememory unit. In a preferred example of this variant, the block of memoryhas a block size that equals a sum of all line lengths of the circulardelay lines. This block of memory is particularly advantageous, since itis possible to store all values, which are stored in the circular delaylines, in the block of memory and vice versa.

In the adaptive filter unit, the number M(f_(n)) of sampling points ofthe first audio signal is equal for all FFT bins. Thus, the calculationsof the processor with respect to the FFT are equal for all FFT bins.This helps to provide equal calculation times for all FFT bins andtherefore calculations, which can be simply performed in parallel.Furthermore, an equal number of sampling points increases acomparability of the received FFT data in the frequency domain.

Alternatively, the number M(f_(n)) of sampling points of the first audiosignal is frequency dependent and not equal for all FFT bins. Theadaptive allows a focusing of the processing resources to those FFTbins, i.e. those frequencies, which are particularly important for theecho cancellation.

According to a second aspect, the invention relates to a hearing device,comprising an audio input interface, a speaker unit, a microphone unit,and the adaptive filter unit, and an audio output interface.

The audio input interface may be configured to receive an audio signal,to convert the audio signal into a first electric audio signal and toprovide the first electric audio signal.

The speaker unit may be configured to receive the first electric audiosignal and to convert the first electric signal into a perceivable audiooutput.

The microphone unit may be arranged and configured to convert anacoustic tone into a second electric audio signal and to provide thesecond electric audio signal.

The adaptive filter unit may bes configured to receive the firstelectric audio signal and the second electric audio signal, and may befurther configured to filter the second electric audio signal based onthe first electric audio signal and to determine a processed audiooutput signal.

The audio output interface may be configured to provide a device outputsignal, which is indicative of the processed audio output signal.

The hearing device has the same advantageous as the adaptive filterunit. In particular, the frequency dependent time delay processed by theadaptive feedback filter allows signal processing resources like amemory of the hearing device to be used in the most important parts ofthe frequency spectrum.

The hearing device according the second aspect of the invention will bedescribed.

The hearing device further comprises a delay controller, arranged andconfigured to receive the first audio signal A(t) and the second audiosignal B(t) and to control the processor by controlling the frequencydependent time delay of the delayed audio estimation data according to alength of an echo that is present in the second audio signal. This canbe done by searching for similar signal characteristics in a respectivefrequency range of the first and second audio signal. The second audiosignal B(t) may not be filtered by the adaptive filter unit, if thedelay controller does not detect an echo in the present first audiosignal, or if a detected echo has an intensity below a predeterminedecho threshold. The delay controller of this variant leads to a fastprocessing of the hearing device.

In the hearing device, the audio input interface and the audio outputinterface are configured to wirelessly communicate with an externalaudio device. Therefore, the hearing device provides a high level ofmobility of the hearing device.

The hearing device may be a headset, a headphone or a hearing aid.

According to a third aspect, the invention relates to a method forperforming echo cancelling. The method comprises the following steps:

-   -   receiving a first electric audio signal, indicative of a first        audio signal A(t);    -   receiving a second electric audio signal, indicative of a second        audio signal B(t);    -   calculating and providing audio estimation data X(f_(n), A(t₁, .        . . , t_(M(fn)))) in the frequency domain by calculating a FFT        transform of the first audio signal A(t), for frequencies        f_(n)=f₁, . . . , f_(N), wherein N is a number of FFT bins, and        with a number of sampling points M(f_(n)) of the first audio        signal A(t);    -   calculating a transformed second audio signal Y(f_(n), B(t)),        formed by a transformation of the second audio signal B(t) into        the frequency domain;    -   calculating a filtered audio signal by subtracting delayed audio        estimation data from the transformed second audio signal,        wherein the delayed audio estimation data comprises a frequency        dependent time delay compared to the transformed second audio        signal; and    -   calculating and providing a processed audio output signal by        transforming the filtered audio signal into the time domain.

The method may further comprising a providing of a delay line structurethat comprises a plurality of delay lines, in particular circular delaylines. Furthermore, the method includes the step of allocating theplurality of delay lines to each frequency f_(n), respectively, whereinthe delay lines vary in line length, which may be defined as a maximumnumber of values that can be stored in the respective delay line.

The filtered audio signal is determined by an adaptive algorithm foreach frequency f_(n). The adaptive algorithm depends on the signalstrength of detected echoes. Furthermore. The adaptive algorithm dependson the signal strength of the second audio signal. In case of a highsignal strength, soft echoes might not need to be filtered. As a resultof this, the processing time of a respective processor can be reduced.

According to a fourth aspect, the invention relates to a computerprogram for controlling an operation of an adaptive filter unit,comprising program code means for causing a processor of the adaptivefilter unit to carry out the method according to the third aspect of theinvention.

The computer, which comprises the computer program for instance form anintegral part of a headset, a headphone or of a hearing aid and can beimplemented as a microcontroller or microprocessor.

BRIEF DESCRIPTION OF DRAWINGS

The aspects of the disclosure may be best understood from the followingdetailed description taken in conjunction with the accompanying figures.The figures are schematic and simplified for clarity, and they just showdetails to improve the understanding of the claims, while other detailsare left out. Throughout, the same reference numerals are used foridentical or corresponding parts. The individual features of each aspectmay each be combined with any or all features of the other aspects.These and other aspects, features and/or technical effect will beapparent from and elucidated with reference to the illustrationsdescribed hereinafter in which:

FIG. 1 illustrates an embodiment of an adaptive filter unit according toa first aspect of the invention;

FIG. 2 illustrates a delay line structure of the embodiment of theadaptive filter unit according to the first aspect of the invention;

FIG. 3 illustrates an embodiment of a hearing device according to asecond aspect of the invention;

FIG. 4 illustrates an embodiment of a method for performing echocancelling according to a third aspect of the invention.

DETAILED DESCRIPTION

The detailed description set forth below in connection with the appendeddrawings is intended as a description of various configurations. Thedetailed description includes specific details for the purpose ofproviding a thorough understanding of various concepts. However, it willbe apparent to those skilled in the art that these concepts may bepracticed without these specific details. Several aspects of theapparatus and methods are described by various blocks, functional units,modules, components, circuits, steps, processes, algorithms, etc.(collectively referred to as “elements”). Depending upon particularapplication, design constraints or other reasons, these elements may beimplemented using electronic hardware, computer program, or anycombination thereof.

The electronic hardware may include microprocessors, microcontrollers,digital signal processors (DSPs), field programmable gate arrays(FPGAs), programmable logic devices (PLDs), gated logic, discretehardware circuits, and other suitable hardware configured to perform thevarious functionality described throughout this disclosure. Computerprogram shall be construed broadly to mean instructions, instructionsets, code, code segments, program code, programs, subprograms, softwaremodules, applications, software applications, software packages,routines, subroutines, objects, executables, threads of execution,procedures, functions, etc., whether referred to as software, firmware,middleware, microcode, hardware description language, or otherwise.

A hearing device may include a hearing aid that is adapted to improve oraugment the hearing capability of a user by receiving an acoustic signalfrom a user's surroundings, generating a corresponding audio signal,possibly modifying the audio signal and providing the possibly modifiedaudio signal as an audible signal to at least one of the user's ears.The “hearing device” may further refer to a device such as an earphoneor a headset adapted to receive an audio signal electronically, possiblymodifying the audio signal and providing the possibly modified audiosignals as an audible signal to at least one of the user's ears. Suchaudible signals may be provided in the form of an acoustic signalradiated into the user's outer ear, or an acoustic signal transferred asmechanical vibrations to the user's inner ears through bone structure ofthe user's head and/or through parts of middle ear of the user orelectric signals transferred directly or indirectly to cochlear nerveand/or to auditory cortex of the user.

The hearing device is adapted to be worn in any known way. This mayinclude i) arranging a unit of the hearing device behind the ear with atube leading air-borne acoustic signals or with a receiver/loudspeakerarranged close to or in the ear canal such as in a Behind-the-Ear typehearing aid or a Receiver-in-the Ear type hearing aid, and/or ii)arranging the hearing device entirely or partly in the pinna and/or inthe ear canal of the user such as in a In-the-Ear type hearing aid orIn-the-Canal/Completely-in-Canal type hearing aid, or iii) arranging aunit of the hearing device attached to a fixture implanted into theskull bone such as in Bone Anchored Hearing Aid or Cochlear Implant, oriv) arranging a unit of the hearing device as an entirely or partlyimplanted unit such as in Bone Anchored Hearing Aid or Cochlear Implant.

A “hearing system” refers to a system comprising one or two hearingdevices, disclosed in present description, and a “binaural hearingsystem” refers to a system comprising two hearing devices where thedevices are adapted to cooperatively provide audible signals to both ofthe user's ears. The hearing system or binaural hearing system mayfurther include auxiliary device(s) that communicates with at least onehearing device, the auxiliary device affecting the operation of thehearing devices and/or benefitting from the functioning of the hearingdevices. A wired or wireless communication link between the at least onehearing device and the auxiliary device is established that allows forexchanging information (e.g. control and status signals, possibly audiosignals) between the at least one hearing device and the auxiliarydevice. Such auxiliary devices may include at least one of remotecontrols, remote microphones, audio gateway devices, mobile phones,public-address systems, car audio systems or music players or acombination thereof. The audio gateway is adapted to receive a multitudeof audio signals such as from an entertainment device like a TV or amusic player, a telephone apparatus like a mobile telephone or acomputer, a PC. The audio gateway is further adapted to select and/orcombine an appropriate one of the received audio signals (or combinationof signals) for transmission to the at least one hearing device. Theremote control is adapted to control functionality and operation of theat least one hearing devices. The function of the remote control may beimplemented in a SmartPhone or other electronic device, theSmartPhone/electronic device possibly running an application thatcontrols functionality of the at least one hearing device.

In general, a hearing device includes i) an input unit such as amicrophone for receiving an acoustic signal from a user's surroundingsand providing a corresponding input audio signal, and/or ii) a receivingunit for electronically receiving an input audio signal. The hearingdevice further includes a signal processing unit for processing theinput audio signal and an output unit for providing an audible signal tothe user in dependence on the processed audio signal.

The input unit may include multiple input microphones, e.g. forproviding direction-dependent audio signal processing. Such directionalmicrophone system is adapted to enhance a target acoustic source among amultitude of acoustic sources in the user's environment. In one aspect,the directional system is adapted to detect (such as adaptively detect)from which direction a particular part of the microphone signaloriginates. This may be achieved by using conventionally known methods.The signal processing unit may include amplifier that is adapted toapply a frequency dependent gain to the input audio signal. The signalprocessing unit may further be adapted to provide other relevantfunctionality such as compression, noise reduction, etc. The output unitmay include an output transducer such as a loudspeaker/receiver forproviding an air-borne acoustic signal transcutaneously orpercutaneously to the skull bone or a vibrator for providing astructure-borne or liquid-borne acoustic signal. In some hearingdevices, the output unit may include one or more output electrodes forproviding the electric signals such as in a Cochlear Implant.

The detailed description set forth below in connection with the appendeddrawings is intended as a description of various configurations. Thedetailed description includes specific details for the purpose ofproviding a thorough understanding of various concepts. However, it willbe apparent to those skilled in the art that these concepts may bepractised without these specific details. Several aspects of theapparatus and methods are described by various blocks, functional units,modules, components, circuits, steps, processes, algorithms, etc.(collectively referred to as “elements”). Depending upon particularapplication, design constraints or other reasons, these elements may beimplemented using electronic hardware, computer program, or anycombination thereof.

The electronic hardware may include microprocessors, microcontrollers,digital signal processors (DSPs), field programmable gate arrays(FPGAs), programmable logic devices (PLDs), gated logic, discretehardware circuits, and other suitable hardware configured to perform thevarious functionality described throughout this disclosure. Computerprogram shall be construed broadly to mean instructions, instructionsets, code, code segments, program code, programs, subprograms, softwaremodules, applications, software applications, software packages,routines, subroutines, objects, executables, threads of execution,procedures, functions, etc., whether referred to as software, firmware,middleware, microcode, hardware description language, or otherwise.

Now referring to FIG. 1, which illustrates a first embodiment of theadaptive filter unit 100.

The adaptive filter unit 100 comprises a first filter input 110, asecond filter input 120, a processor 130 and a filter output 150.

The first filter input 110 is configured to receive a first electricaudio signal 115, indicative of a first audio signal 118 A(t).

The second filter input 120 is configured to receive a second electricaudio signal 125, indicative of a second audio signal 128 B(t). In theshown embodiment, the first and second audio signal 118, 128 areprovided digitally and therefore comprise an intensity information ofthe respective audio signal for a discrete number of times.

The processor 130 is connected to the first and second filter input 110,120 and arranged and configured to receive the first and second electricaudio signal 118, 128. After receiving the first audio signal 118, theprocessor is configured to calculate and provide audio estimation data132 X(f_(n), A(t₁, . . . , t_(M(fn)))) in the frequency domain bycalculating a FFT transform of the first audio signal A(t), forfrequencies f_(n)=f₁, . . . , f_(N), wherein N is a number of FFT bins,and with a number of sampling points M(f_(n)) of the first audio signalA(t). This calculation is carried out by a first FFT unit 134. Theprocessor 130 is further configured to calculate a transformed secondaudio signal 136 Y(f_(n), B(t)), formed by a transformation of thesecond audio signal 128 B(t) into the frequency domain, in a second FFTunit 138. Delayed audio estimation data 142 is subtracted from thetransformed second audio signal 136 by the processor 130 in asubtracting unit 140. Thereby, the subtraction unit 140 calculates afiltered audio signal 143. The delayed audio estimation data 142 isprovided by a memory unit 144 of the adaptive filter unit 100, which isarranged to provide a data exchange 146 with the processor 130. The dataexchange further comprises a providing of the audio estimation data 132to the memory unit 144. The delayed audio estimation data 142 comprisesa frequency dependent time delay compared to the transformed secondaudio signal 136. A reverse FFT unit 148 of the processor 130 is furtherconfigured to calculate a processed audio output signal 149 bytransforming the filtered audio signal 143 into the time domain. Thedetailed calculation schemes of the first, the second and the reverseFFT units 134, 138, 148 are provided as standard FFT schemes, which arewell known in the art.

The filter output 150 is configured to provide the processed audiooutput signal 155.

The memory unit 144 of this embodiment comprises a plurality of circulardelay lines allocated to each frequency f_(n), respectively, wherein thecircular delay lines vary in line length, which is defined as a maximumnumber of values that can be stored in the respective circular delayline. The structure of the circular delay lines and of the line lengthsof this embodiment are illustrated and discussed in the course of FIG.2.

For reasons of simplicity, it is not illustrated that the memory unit144 of this embodiment further comprises a base address registerallocating each circular delay line to a base address, and furthercomprises separate bookkeeping buffers, which are configured to storethe line length and the base address of each circular delay line. Thisallows the memory of the memory unit 144 to be formed as a block ofmemory that stores the base address and the line length of each delayline wherever memory is free for storing data. The not shown block ofmemory has a block size that equals the sum of all delay lines in orderto store all values that are stored within the circular delay lines ateach point in time.

FIG. 2 illustrates a delay line structure 200 of the embodiment of theadaptive filter unit 100 shown in FIG. 1.

The axis of abscissas 210 shows the FFT bins 215 in the frequencydomain. The frequencies f_(n)=f₁, . . . , f_(N), for which the FFTtransform is calculated by the processor 130, are the respective centerfrequencies 218 of each FFT bin 215. All FFT bins 215 of this embodimenthave the same size meaning that they all cover the same frequency range.

The axis of ordinates 220 shows the line length of each delay line 225allocated to a respective FFT bin 215, i.e. to a center frequency 218.The delay lines 225 are circular delay lines, so that each value storedin the delay line is overwritten after a predefined period of time,which depends on the line length. Furthermore, the circular delay linesof this embodiment vary in line length, as illustrated in FIG. 2.

A single grayscale delay line frame 230 is shown, which comprises theaudio estimation data X(f_(n), A(t′₁, . . . , t′_(M(fn)))) for those FFTbins 215, which have a line length of at least L_(FR). The points intime t′₁, . . . , t′_(M(fn)) are thus before the time t₁ of the mostrecent delay line frame 235. The delay line frames that have a linelength of more than L_(FR) comprise audio estimation data that has beenrecorded before the times t′₁, . . . t′_(M(fn)).

Therefore, FIG. 2 illustrates that the circular delay lines 225 with thelargest line length are allocated to the FFT bins 215 with small centerfrequencies 218. Thus, the adaptive filter unit 100 allows the filteringof long echoes in the low frequency area by subtracting the respectivestrongly delayed audio estimation data from the transformed second audiosignal 136.

In order to avoid wasting memory the line lengths may be limited topowers of two.

FIG. 3 illustrates an embodiment of a hearing device 300.

The hearing device 300 comprises an audio input interface 310, a speakerunit 320, a microphone unit 330, the adaptive filter unit 100 accordingto an embodiment of the first aspect of the invention, and an audiooutput interface 340.

The audio input interface 310 is configured to receive an audio signal315, to convert the audio signal 315 into the first electric audiosignal 115 and to provide the first electric audio signal 115.

The speaker unit 320 is configured to receive the first electric audiosignal 115 and to convert the first electric signal 115 into aperceivable audio output 325.

The microphone unit 330 is arranged and configured to convert anacoustic tone 335 into the second electric audio signal 125 and toprovide the second electric audio signal 125. The acoustic tone 335typically comprises at least parts of the audio output 325, which havebeen reflected by the environment of the hearing device 300 and thusform an echo 338.

The adaptive filter unit 100 is configured to receive the first electricaudio signal 115 and the second electric audio signal 125, and isfurther configured to filter the second electric audio signal 125 basedon the first electric audio signal 115 and to determine a processedaudio output signal 155.

The audio output interface 340 is configured to provide a device outputsignal 345, indicative of the processed audio output signal 155.

The audio input interface 310 and the audio output interface 340 of theembodiment shown in FIG. 3 are configured to wirelessly communicate withan external audio device (not shown). Both interfaces 310, 340 thuscomprise an antenna system (not shown) for receiving and/or transmittingwireless signals 315, 345. The external audio device is in thisembodiment a mobile phone, a notebook, a headphone, a headset or ahearing aid of a further user, who communicates with the user of thehearing device 300.

The illustrated hearing device 300 is a headset. However, in embodimentsnot shown, the hearing device is a headphone, a hearing aid or anotherdevice with at least one microphone and at least one speaker foroutputting and receiving acoustics tones.

In an embodiment not shown, the hearing device further comprises a delaycontroller, arranged and configured to receive the first audio signalA(t) and the second audio signal B(t) and to control the processor bycontrolling the frequency dependent time delay of the delayed audioestimation data according to a length of an echo that is present in thesecond audio signal. The delay controller of this embodiment is arrangedwithin a signal path between the microphone unit and the adaptive filterunit and directly connected to the processor of the adaptive filterunit. Upon detection of similar signal characteristics in a respectivefrequency range of the first and second audio signal, i.e. of an echo,the delay controller is configured to activate the filtering of thesecond audio signal by the adaptive filter unit. In further embodimentsnot shown, the delay controller is further configured to determine,whether the detected echo has an intensity below or above apredetermined echo threshold and to activate the filtering only if theintensity of the echo is above the echo threshold.

FIG. 4 illustrates an embodiment of a method 400 for performing echocancelling. The method 400 comprises steps 410, 420, 430, 440, 450, 460that are given in the following.

The first step 410 is formed by a reception of a first electric audiosignal, which is indicative of a first audio signal A(t).

In the further step 420, a second electric audio signal, which isindicative of a second audio signal B(t), is received.

The next step 430 includes a calculation and a provision of an audioestimation data X(f_(n), A(t₁, . . . , t_(M(fn)))) in the frequencydomain by calculating a FFT transform of the first audio signal A(t),for frequencies f_(n)=f₁, . . . , f_(N), wherein N is a number of FFTbins, and with a number of sampling points M(f_(n)) of the first audiosignal A(t).

Afterwards, a transformed second audio signal Y(f_(n), B(t)), formed bya transformation of the second audio signal B(t) into the frequencydomain, is calculated in the step 440.

In another step 450, a filtered audio signal is calculated bysubtracting delayed audio estimation data from the transformed secondaudio signal, wherein the delayed audio estimation data comprises afrequency dependent time delay compared to the transformed second audiosignal.

The last step 460 of the method 400 is formed by calculating andproviding a processed audio output signal by transforming the filteredaudio signal into the time domain.

The illustrated order of the steps 410, 420, 430, 440, 450, 460 in thisembodiment forms the preferred order of the method according to thethird aspect of the invention.

As used, the singular forms “a,” “an,” and “the” are intended to includethe plural forms as well (i.e. to have the meaning “at least one”),unless expressly stated otherwise. It will be further understood thatthe terms “includes,” “comprises,” “including,” and/or “comprising,”when used in this specification, specify the presence of statedfeatures, integers, steps, operations, elements, and/or components, butdo not preclude the presence or addition of one or more other features,integers, steps, operations, elements, components, and/or groupsthereof. It will also be understood that when an element is referred toas being “connected” or “coupled” to another element, it can be directlyconnected or coupled to the other element but an intervening elementsmay also be present, unless expressly stated otherwise. Furthermore,“connected” or “coupled” as used herein may include wirelessly connectedor coupled. As used herein, the term “and/or” includes any and allcombinations of one or more of the associated listed items. The steps ofany disclosed method is not limited to the exact order stated herein,unless expressly stated otherwise.

It should be appreciated that reference throughout this specification to“one embodiment” or “an embodiment” or “an aspect” or features includedas “may” means that a particular feature, structure or characteristicdescribed in connection with the embodiment is included in at least oneembodiment of the disclosure. Furthermore, the particular features,structures or characteristics may be combined as suitable in one or moreembodiments of the disclosure. The previous description is provided toenable any person skilled in the art to practice the various aspectsdescribed herein. Various modifications to these aspects will be readilyapparent to those skilled in the art, and the generic principles definedherein may be applied to other aspects.

The claims are not intended to be limited to the aspects shown herein,but is to be accorded the full scope consistent with the language of theclaims, wherein reference to an element in the singular is not intendedto mean “one and only one” unless specifically so stated, but rather“one or more.” Unless specifically stated otherwise, the term “some”refers to one or more.

REFERENCE NUMBER LIST

-   100 adaptive filter unit-   110 first filter input-   115 first electric audio signal-   118 first audio signal-   120 second filter input-   125 second electric audio signal-   128 second audio signal-   130 processor-   132 audio estimation data-   134 first FFT unit-   136 transformed second audio signal-   138 second FFT unit-   140 subtracting unit-   142 delayed audio estimation data-   143 filtered audio signal-   144 memory unit-   146 data exchange-   148 reverse FFT unit-   149 audio output signal-   150 filter output-   155 processed audio output signal-   200 delay line structure-   210 axis of abscissas-   215 FFT bin-   218 center frequency-   220 axis of ordinates-   225 delay line-   230 delay line frame-   235 most recent delay line frame-   300 hearing device-   310 audio input interface-   315 audio signal-   320 speaker unit-   325 audio output-   330 microphone unit-   335 acoustic tone-   338 echo-   340 audio output interface-   345 device output signal-   400 method-   410, 420, 430 440, 450, 460 steps of the method

1. An adaptive filter unit, in particular for being used as an echocanceller, comprising a first filter input, configured to receive afirst electric audio signal, indicative of a first audio signal A(t); asecond filter input, configured to receive a second electric audiosignal, indicative of a second audio signal B(t); a processor, arrangedand configured to receive the first and second electric audio signal;calculate and provide audio estimation data X(f_(n), A(t₁, . . . ,t_(M(fn)))) in the frequency domain by calculating a FFT transform ofthe first audio signal A(t), for frequencies f_(n)=f₁, . . . , f_(N),wherein N is a number of FFT bins, and with a number of sampling pointsM(f_(n)) of the first audio signal A(t); calculate a transformed secondaudio signal Y(f_(n), B(t)), formed by a transformation of the secondaudio signal B(t) into the frequency domain; calculate a filtered audiosignal by subtracting delayed audio estimation data from the transformedsecond audio signal, wherein the delayed audio estimation data isprovided by a memory unit of the adaptive filter unit, which is arrangedto provide a data exchange with the processor, and wherein the delayedaudio estimation data comprises a frequency dependent time delaycompared to the transformed second audio signal; calculate a processedaudio output signal by transforming the filtered audio signal into thetime domain; and a filter output, configured to provide the processedaudio output signal.
 2. The adaptive filter according to claim 1,wherein the processor comprises a multidelay filter structure.
 3. Theadaptive filter unit according to claim 1, wherein the memory unitcomprises a delay line structure comprising at least one delay line,which contains at least the audio estimation data X(f_(n), A(t′₁, . . ., t′_(M(fn)))) for points in time t′₁, . . . , t′_(M(fn)) being beforethe time t₁.
 4. The adaptive filter unit according to claim 3, whereinthe memory unit comprises a plurality of circular delay lines allocatedto each frequency f_(n), respectively, wherein the circular delay linesvary in line length, which is defined as a maximum number of values thatcan be stored in the respective circular delay line.
 5. The adaptivefilter unit according to claim 1, wherein the data exchange between theprocessor and the memory unit comprises an allocation of a value of theaudio estimation data X(f_(n), A(t₁, . . . , t_(M(fn)))) for eachfrequency f_(n) to a memory address of a memory structure of the memoryunit.
 6. The adaptive filter unit according to claim 3, wherein eachdelay line comprises a line length that is limited to powers of two. 7.The adaptive filter unit according to claim 5, wherein the memory unitcomprises a base address register allocating each circular delay line toa base address, and further comprises at least one bookkeeping buffer,configured to store the line length and the base address of eachcircular delay line.
 8. The adaptive filter unit according to claim 1,wherein the number M(f_(n)) of sampling points of the first audio signalis equal for all FFT bins.
 9. A hearing device, comprising an audioinput interface, configured to receive an audio signal, to convert theaudio signal into a first electric audio signal and to provide the firstelectric audio signal; a speaker unit, configured to receive the firstelectric audio signal and to convert the first electric signal into aperceivable audio output; a microphone unit, arranged and configured toconvert an acoustic tone into a second electric audio signal and toprovide the second electric audio signal; an adaptive filter unitaccording to claim 1, configured to receive the first electric audiosignal and the second electric audio signal, and which is furtherconfigured to filter the second electric audio signal based on the firstelectric audio signal and to determine a processed audio output signal;and an audio output interface, configured to provide a device outputsignal, indicative of the processed audio output signal.
 10. The hearingdevice according to claim 9, further comprising a delay controller,arranged and configured to receive the first audio signal A(t) and thesecond audio signal B(t) and to control the processor by controlling thefrequency dependent time delay of the delayed audio estimation dataaccording to a length of an echo that is present in the second audiosignal.
 11. The hearing device according to claim 9, wherein the audioinput interface and the audio output interface are configured towirelessly communicate with an external audio device.
 12. The hearingdevice according to claim 9, wherein the hearing device is a headset, aheadphone or a hearing aid.
 13. A method for performing echo cancelling,comprising the steps of receiving a first electric audio signal,indicative of a first audio signal A(t); receiving a second electricaudio signal, indicative of a second audio signal B(t); calculating andproviding audio estimation data X(f_(n), A(t₁, . . . , t_(M(fn)))) inthe frequency domain by calculating a FFT transform of the first audiosignal A(t), for frequencies f_(n)=f₁, . . . , f_(N), wherein N is anumber of FFT bins, and with a number of sampling points M(f_(n)) of thefirst audio signal A(t); calculating a transformed second audio signalY(f_(n), B(t)), formed by a transformation of the second audio signalB(t) into the frequency domain; calculating a filtered audio signal bysubtracting delayed audio estimation data from the transformed secondaudio signal, wherein the delayed audio estimation data comprises afrequency dependent time delay compared to the transformed second audiosignal; and calculating and providing a processed audio output signal bytransforming the filtered audio signal into the time domain.
 14. Themethod according to claim 13, further comprising providing a delay linestructure comprising a plurality of delay lines, in particular circulardelay lines; allocating the plurality of delay lines to each frequencyf_(n), respectively, wherein the delay lines vary in line length, whichis defined as a maximum number of values that can be stored in therespective delay line.
 15. The method according to claim 13, wherein thefiltered audio signal is determined by an adaptive algorithm for eachfrequency f_(n).
 16. A non-transitory computer-readable medium storing aprogram for controlling an operation of an adaptive filter unit,comprising program code means for causing a processor of the adaptivefilter unit to carry out a method according to claim
 13. 17. Theadaptive filter unit according to claim 2, wherein the memory unitcomprises a delay line structure comprising at least one delay line,which contains at least the audio estimation data X(f_(n), A(t′₁, . . ., t′_(M(fn)))) for points in time t′₁, . . . , t′_(M(fn)) being beforethe time t₁.
 18. The adaptive filter unit according to claim 2, whereinthe data exchange between the processor and the memory unit comprises anallocation of a value of the audio estimation data X(f_(n), A(t₁, . . ., t_(M(fn)))) for each frequency f_(n) to a memory address of a memorystructure of the memory unit.
 19. The adaptive filter unit according toclaim 3, wherein the data exchange between the processor and the memoryunit comprises an allocation of a value of the audio estimation dataX(f_(n), A(t₁, . . . , t_(M(fn)))) for each frequency f_(n) to a memoryaddress of a memory structure of the memory unit.
 20. The adaptivefilter unit according to claim 4, wherein the data exchange between theprocessor and the memory unit comprises an allocation of a value of theaudio estimation data X(f_(n), A(t₁, . . . , t_(M(fn)))) for eachfrequency f_(n) to a memory address of a memory structure of the memoryunit.